RTCRtpEncodingParameters
E134377
RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
All labels observed (1)
| Label | Occurrences |
|---|---|
| RTCRtpEncodingParameters canonical | 4 |
How this entity was disambiguated
This entity first appeared as the object of triple T1166455 — resolving that mention is where its identity was fixed. The disambiguator weighed these candidate entities and picked the highlighted one (or “None”, minting a new entity). This is how homonymy is resolved: the same surface form can point to different entities.
NED1
Entity disambiguation (via context triple)
gpt-5-mini-2025-08-07
Target entity: RTCRtpEncodingParameters Context triple: [RTCRtpReceiver, relatedTo, RTCRtpEncodingParameters]
-
A.
RTCRtpReceiver
RTCRtpReceiver is a WebRTC API interface that represents the receiving end of an RTP media stream, providing access to incoming audio or video tracks and related reception statistics.
-
B.
SRTP for secure media transport
SRTP for secure media transport is a security protocol that encrypts and authenticates real-time audio and video streams, commonly used in VoIP and WebRTC communications.
-
C.
G.722
G.722 is a wideband audio codec standard that provides higher-quality voice transmission than traditional narrowband codecs, commonly used in VoIP and teleconferencing applications.
-
D.
G.729
G.729 is a widely used ITU-T audio codec standard that compresses voice for bandwidth-efficient transmission in VoIP and other telephony applications.
-
E.
G.711
G.711 is a widely used ITU-T audio codec standard for pulse code modulation (PCM) of voice in traditional telephony and VoIP systems.
- F. None of above. chosen
- G. Unsure - the case is ambiguous/there is not enough information to decide.
NED2
Entity disambiguation (via description)
gpt-5-mini-2025-08-07
Target entity: RTCRtpEncodingParameters Target entity description: RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
-
A.
RTCRtpReceiver
RTCRtpReceiver is a WebRTC API interface that represents the receiving end of an RTP media stream, providing access to incoming audio or video tracks and related reception statistics.
-
B.
SRTP for secure media transport
SRTP for secure media transport is a security protocol that encrypts and authenticates real-time audio and video streams, commonly used in VoIP and WebRTC communications.
-
C.
G.722
G.722 is a wideband audio codec standard that provides higher-quality voice transmission than traditional narrowband codecs, commonly used in VoIP and teleconferencing applications.
-
D.
G.729
G.729 is a widely used ITU-T audio codec standard that compresses voice for bandwidth-efficient transmission in VoIP and other telephony applications.
-
E.
G.711
G.711 is a widely used ITU-T audio codec standard for pulse code modulation (PCM) of voice in traditional telephony and VoIP systems.
- F. None of above. chosen
Statements (50)
| Predicate | Object |
|---|---|
| instanceOf |
WebRTC configuration object
ⓘ
dictionary type ⓘ |
| appliesTo |
audio RTP streams
ⓘ
video RTP streams ⓘ |
| controls | encoding of a particular RTP stream ⓘ |
| definedIn |
WebRTC specification
ⓘ
linked to:
WebRTC
|
| hasProperty |
active
ⓘ
adaptivePtime ⓘ codecPayloadType ⓘ dtx ⓘ fec ⓘ maxBitrate ⓘ maxFramerate ⓘ maxptime ⓘ networkPriority ⓘ priority ⓘ ptime ⓘ rid ⓘ rtx ⓘ scaleResolutionDownBy ⓘ scaleResolutionDownTo ⓘ ssrc ⓘ |
| hasPurpose |
configure RTP encoding for a media stream
ⓘ
configure redundancy and error resilience ⓘ configure scalability of RTP streams ⓘ control bitrate of RTP streams ⓘ |
| partOf |
WebRTC
ⓘ
WebRTC RTCRtpSender API ⓘ
linked to:
RTCRtpSender
|
| propertyType_active | boolean ⓘ |
| propertyType_codecPayloadType | byte ⓘ |
| propertyType_dtx | RTCDtxStatus ⓘ |
| propertyType_fec | RTCRtpFecParameters ⓘ |
| propertyType_maxBitrate | unsigned long ⓘ |
| propertyType_maxFramerate | double ⓘ |
| propertyType_maxptime | unsigned long ⓘ |
| propertyType_networkPriority | RTCPriorityType ⓘ |
| propertyType_priority | RTCPriorityType ⓘ |
| propertyType_ptime | unsigned long ⓘ |
| propertyType_rid | DOMString ⓘ |
| propertyType_rtx | RTCRtpRtxParameters ⓘ |
| propertyType_scaleResolutionDownBy | double ⓘ |
| propertyType_ssrc | unsigned long ⓘ |
| relatedTo |
RTCRtpCodecParameters
ⓘ
RTCRtpHeaderExtensionParameters ⓘ |
| supports |
scalable video coding configurations
ⓘ
simulcast configurations ⓘ |
| usedBy |
RTCRtpSendParameters
ⓘ
RTCRtpSender ⓘ |
| usedInMethod |
RTCRtpSender.getParameters()
ⓘ
RTCRtpSender.setParameters() ⓘ |
How these facts were elicited
The pipeline generated the facts above by prompting gpt-5.1 with this entity's name + description and the instruction below.
Instruction
You are a knowledge base construction expert. Given a subject entity and a description of it, return factual statements that you know for the subject as a JSON list of dictionaries(triples), where keys must be "subject", "predicate" and "object". The number of facts may be very high, between 25 to 50 or more, for very popular subjects. For less popular subjects, the number of facts can be very low, like 5 or 10. # Requirements - If you don't know the subject at all, return an empty list. - If the subject is not a named entity, return an empty list. - Include at least one triple where predicate is "instanceOf". - Do not get too wordy. - Separate several objects into multiple triples with one object.
Input
Subject: RTCRtpEncodingParameters Description of subject: RTCRtpEncodingParameters is a WebRTC configuration object that defines how a particular RTP stream is encoded, including settings like bitrate, scalability, and redundancy for real-time audio or video transmission.
Referenced by (4)
Full triples — surface form annotated when it differs from this entity's canonical label.